1. Field of the Invention
This invention relates to digital apparatus and a method for compressing analog or digital data for transmission by standard commercial data links using adaptive transform coding (ATC) to achieve an optimal allocation of the compressed digital bits to be transmitted. More specifically, the present invention relates to hardware for achieving fast allocation of bits to define blocks of information which permit real time data compression and transmission.
2. Description of the Prior Art
Digital signalling formats are superior to traditional analog formats in many ways. As a result there is a trend to convert analog telephone networks to all digital networks. High grade speech occupies a bandwidth of about 3200 hertz in analog form. A present way of converting this analog speech to digital format is to sample the speech in blocks of 8000 samples per second and assign a digital value to each sample using eight bits per sample. This typical conversion requires a bandwidth which will support transmission of 64 Kbps to reproduce a speech waveform with telephone quality when decoded. This is a most important reason that standard digital data-telephone phone links have been standardized at 64 Kbps data rates.
Standard digital speech phone lines capable of transmitting sixty-four thousand bits per second are commonly used for telephone voice communication. When digital telephone lines are employed, all 64 Kbps capacity is available to the user to use in any manner desirable. It is possible to use such a standard telephone data link line to transmit up to eight voice channels digitally on the same data link line with understandable, though somewhat degraded, voice quality. It is also possible to compress the digital data representations by a ratio of four to one and transmit four voice channels on one digital data speech phone line with very little voice degradation. Heretofore, the delay in achieving such speech compression has mitigated against its use for real time transmission.
Transcontinental digital speech phone lines rent or lease for over $40,000 per month. If the voice information to be transmitted is compressed at a ratio of four to one, the user will achieve saving of over $120,000 per month.
For over twenty-five years scholarly articles have been presented proposing different ways of compressing voice information, however, there is not presently available commercially any apparatus to take advantage of the aforementioned possible savings when transmitting compressed voice information.
Huang and Schulthiess in their paper "Block Quantization of Correlated Gaussian Random Variables", IEEE Tran.-Comm. Sys. vol. CS11, pp. 286-296, September 1963, suggested that blocks of digital information could be encoded on a sample by sample basis and further suggested that digital bit allocation could be employed to define the block of information to achieve data compression. The allocation of bits suggested by this paper is sub-optimal because the optimal distribution of bits b(j) is not a linear function of the variance .sigma..sup.2 (j) and requires a noninteger assignment of bits which rarely results in a real distribution of the available bits.
Adrian Segall in his paper "Bit Allocation and Encoding for Vector Sources", IEEE Trans Inform Theory vol. IT-22, pp. 162-169, March 1976, reviewed the different ways of allocating digital bits to achieve data compression of blocks of digital information. Further, this Segall article suggested an algorithm for the optimum allocation of discrete bits to achieve data compression of blocks of digital information. While this paper suggests an optimum allocation of bits, the implementation would require approximately sixty thousand separate computer operations. To perform the same optimum allocation of bits the present invention only requires approximately 7,560 operations which can be performed as simple operations in real time.
Richard V. Cox and Ronald Crochiere of Bell Labs in their paper "Real Time Simulation of Adaptive Transform Coding", IEEE Trans. on Acoustic, Speech and Signal Processing, vol. ASSP 29, No. 2, April 1981 suggested Adaptive Transform Coding (ATC) as an effective means of digitally encoding speech at low bit rates (9.6-16 Kpbs) for transmission on voice lines at 64 Kbps.
Actual telephone conversations were data compressed and transform encoded. The encoded data was transmitted and then transform decoded from the compressed form at the receiving end. This Cox et al article further teaches an ATC encoder and decoder employing a fast processor intended to enable real time data compression transmission.
It would be extremely desirable to provide in a single hardware package a system which would transmit and/or receive two, four or more channels of voice information on a single standard commercial data link channel without noticeable distortion.